[Asterisk-Users] SIP UDP ports on firewal to open
I notice most things say to open ports 1-2 for UDP for SIP, however from time to time this range isn't where Asterisk is opening the ports: We're at xxx.xxx.xxx.xxx port 8542 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) This call has no audio, presumably because port 8542 is firewalled in the iptables on the server. Is there somewhere I'm missing to port a port range in sip.conf (shouldn't there be if there isn't?). Steve Radich BitShop, Inc. - http://www.bitshop.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP UDP ports on firewal to open
Hallo Steve Radich On Tue, 25 Jan 2005 11:48:44 -0500 you wrote: I notice most things say to open ports 1-2 for UDP for SIP, however from time to time this range isn't where Asterisk is opening the ports: We're at xxx.xxx.xxx.xxx port 8542 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) This call has no audio, presumably because port 8542 is firewalled in the iptables on the server. Is there somewhere I'm missing to port a port range in sip.conf (shouldn't there be if there isn't?). Well you should ask your (sip)provider which ports he is going to use :) Steve Radich BitShop, Inc. - http://www.bitshop.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tho/\/\as ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP UDP ports on firewal to open
I notice most things say to open ports 1-2 for UDP for SIP, however from time to time this range isn't where Asterisk is opening the ports: We're at xxx.xxx.xxx.xxx port 8542 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) This call has no audio, presumably because port 8542 is firewalled in the iptables on the server. Is there somewhere I'm missing to port a port range in sip.conf (shouldn't there be if there isn't?). Each sip device (including *) has a udp port range set by the OEM. For *, take a look at rtp.conf and you'll see 10,000 to 20,000. For Cisco phones, look in SIPDefault.cnf and you'll see they use 16,384 to 32,766. So, data flows from * to Cisco might use udp 10,123 but Cisco to * might use 31,123 for rtp flows. Unless you change the port range for every sip phone coming through your firewall to some consistent range, expect to see whatever the OEM decided to use for their product. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
